>> What do you mean exactly by "just works"? Are the external phones >> supposed to talk with the internal phones? > > Not directly, they go through the server
I'm guessing only the SIP signalling goes through the Asterisk server, and not the RTP media (i.e. you don't do any kind of media anchoring). >> Do the internal phones have >> public or private addresses? > > Private interface so private address Since I don't have a clear picture of your network I will assume that the internal phones go through NAT if they need to get their media streams out to the Internet. Therefore, if the internal phones are to perform VoIP calls with external phones, how are you going to deal with the private IPs being present in the SDP payload on SIP INVITEs and 200 OKs? The internal phones may be able to send their RTP streams to the public phones, but the public phones won't be able to send their audio stream back to the private addresses referenced in the private's phone SDP payload. Therefore I think you'll end up with one way audio only.. unless you anchor/proxy all media as well on the Asterisk (I don't know Asterisk so I don't know if it does that). I'm more familiar with the IMS domain where Session Border Gateways perform the functions of firewall/policy enforcer/QoS/media anchoring and address the issues inherent when using SIP throught NAT/firewall. -Martin
